Voice over IP architecture

ABSTRACT

A Voice over Internet Protocol (VoIP) network is described in which session state is maintained in access switches, but not signaling gateways which maintain transaction state only during pendency of a related transaction. The signaling gateway further provides transparent inter-operation between the VoIP network and non-IP networks, such as the PSTN, by means of a translator which directly translates messages between the networks.

CROSS-REFERENCE TO RELATED APPLICATIONS

The present application is a divisional application of U.S. applicationSer. No. 10/071,088, filed Feb. 11, 2002, now U.S. Pat. No. 7,139,263,issued Nov. 21, 2006, which is incorporated herein by reference in itsentirety, which claims the benefit under 35 U.S.C. §119(e) of U.S.Provisional Patent Application No. 60/330,330 filed Oct. 19, 2001,entitled “Transaction Oriented Signaling Gateway”.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates to telecommunications systemarchitectures, and more specifically to Voice over Internet Protocol(VOIP) architectures. Improved access methodologies and systemcomponents, including an access switch and signaling gateway, aredescribed within the context of a novel VoIP architecture. Theseimproved system components and methodologies manage session state moreefficiently and provide a clear migration path from conventional systemarchitectures.

2. Description of the Related Art

Although VoIP has been in existence for many years, evolving servicedemands are forcing a rapid evolution of VoIP technology. The pace atwhich new services are being integrated into existing networks continuesto increase as VoIP products and services develop. Though stillevolving, packet-based telephony is becoming increasingly sophisticated.Indeed, voice protocols have developed to offer a rich set of features,scalability, and standardization than what was unavailable only a fewyears ago.

Critical to success of any telecommunications system is the ability todeploy value-added and high margin services. For many reasons, a few ofwhich are discussed below, VoIP and other IP-based technologies are bestpositioned to realize these profitable services.

Economics is perhaps the single greatest motivation for the developmentand deployment of VoIP. Increased competition among existing andemerging voice service vendors has brought tremendous downward pressureon the cost of voice services in the telecommunications market. Thistrend is likely to continue with further declines in the price of voiceservices.

Additionally, many existing networks support mostly data (Internet)services that are based on IP. Many service providers already own andwill continue to build-out IP infrastructures. New service providerswishing to enter the voice services market will almost certainlytransport voice traffic across the existing IP backbone. Building aparallel voice services network based on legacy circuit-switchingequipment is simply not a cost-effective option.

Many emerging competitive local exchange carriers (CLECs) are sensitiveto the cost of developing voice service networks. For many, the cost oflegacy voice circuit-switching equipment is prohibitively high. Also thecost of the space, personnel, and operations capabilities required tomaintain such networks is prohibitive. These carriers need a networkthat can be leveraged to realize other data services, such as Internet,virtual private networks, and managed network offerings. In order toprovision such services and provide voice services, VoIP is an idealsolution.

Major telecommunications carriers are looking for ways to cut the costof running and upgrading existing voice networks. These carriers want toreplace and augment their existing networks with VoIP solutions for manyof the reasons described above. Additionally, VoIP offers the carriers apath to circumvent existing tariff regulations. That is, carriers mayuse data services to transport voice calls to get around traditional(regulated) pricing structures and reduce the total cost of voiceservices.

In addition to cost advantages, VoIP networks offer compelling technicaladvantages over circuit switching networks. VoIP networks are more opento technical improvement and competition-drive improvement thancircuit-switching networks which are dominated by entrenched equipmentvendors. The open, standards-based architecture provided by VoIPnetworks allows greater interchangeability and more modularity thanproprietary, monolithic, circuit-switching networks. Open standards alsotranslate into the realization of new services that are rapidlydeveloped and deployed, rather than waiting for a particular vendor todevelop a proprietary solution.

In sum, the promise offered by VoIP technology is just beginning to berealized. As will be discussed hereafter, the contemporary state of VoIPtechnology and architecture is not without limitations anddisadvantages. Yet, the utility offered by VoIP networks, as compared totraditional circuit-switching networks, is immense and beyond question.The issue is really one of optimally defining a VoIP architecture andproviding enabling system components and access methodologies.

The advantages and preferred implementation of the present invention arebest viewed and understood against the backdrop of the contemporary VoIParchitectures. However, an understanding of contemporary VoIP technologyrequires at least some understanding of the public switched telephonenetwork (PSTN).

There are four major tasks performed by the PSTN to connect a call.While the PSTN is capable of providing other services beyondpoint-to-point voice calls such services are predicated upon thefollowing basic tasks: (1) signaling; (2) database services; (3) callset-up and tear-down; and (4) analog voice to digital data conversion.

Phones calls are inherently connection oriented. That is, a connectionto the called entity must be established ahead of time before aconversation can occur. Switches, the central components in the PSTN,are responsible for creating this connection. Between the circuitswitches are connections (trunk lines) that carry the voice traffic.These links vary in data communication speed from T-1 and E-1 toOC-192/STM-64, with individual channels (DS-0s) in each link typerepresenting one voice channel. Switches are also responsible forconverting the analog voice signal into a digital data format that maybe transported across the network.

Signaling notifies both the network and its users of important events.Examples of signaling range from telephone ringer activation to thedialing of digits used to identify a called entity. Network elementsalso use signaling to create connections through the network.

The Signaling System Seven (SS7) is a standard, packet-based networkthat transports signaling traffic between the switches involved in thecall. FIG. 1 illustrates the basic flow of a telephone call through thePSTN and SS7 network. A call begins when a user dials a destinationphone number using call origination equipment 1. A local switch (notshown) typically analyses the destination number to determine if thecall is a local one. If the call is local, the local switch may directlyconnect the call. More typically, the dialed telephone number results ina query to one or more service control points (SCP) 3. SCPs aredatabases that execute queries and translate the dialed telephone phonenumbers into a set of circuit switching commands. SCPs also allow suchcommon telephone features as 800 number support, 911 service, and caller10. Signaling switch points (SSPs, not shown) are the interface betweenswitches 4 and the SS7 network. It is here that SS7 messages aretranslated into the connection details required by the switches tophysically connect the call origination point to the call destinationpoint.

The SS7 control network is “out of band,” that is, it is not transmittedwithin the same links used to carry the actual voice channels.Specialized equipment called Signal Transfer Points (STPs) 6 transportthe SS7 signaling messages. As will be seen hereafter, these STPs areanalogous to IP routers in that the messages are carried in data packetscalled message transfer parts.

The SS7 network is very expansive and has been built up over manydecades of effort and expense. Indeed, the SS7 network is actually acollection of signaling networks deployed throughout the developedworld. There are many technical and historical reasons why the signalingportion of the network is broken out from the rest of the system.However, the greatest value in such a design is the ability to addnetwork intelligence and features without a dependency on the underlyingcircuit-switching infrastructure.

Ultimately, the destination switch signals the destination equipment 5that a call has arrived, typically by activating a ringer. When thecalled party picks up the receiver and completes the circuit, aconversation may take place. Throughout the conversation, switchesconvert analog voice signals into digital data capable of beingtransported over the network.

Once the call is complete, the switches notify the SS7 network which“tears-down” the circuit connections. In contrast to the “setup”function that identifies and enables the collection of circuitconnections that facilitate the call, the tear-down function disablesthe connections, thus freeing up the switching resources for subsequentuse by the network. As one would suspect there are many more detailsinvolved in making a telephone call, but these steps describe the basicflow of events. For example, a great many supervisory messages arecommunicated via the SS7 network during any one completed call.

Of particular note are the concepts of “call state” and “call control.”Call state is a general term referring to a great quantity of data thatis maintained within the network. Call state may be stored at numerouspoints within the network. It includes at least data identifying theorigination point of a call, the destination point for the call, and theswitching control data required to connect the two points. Call statemay also include data identifying particular mechanisms being used toimplement and transport the call, such as analog-to-digital conversiontype, echo cancellation parameters, noise elimination techniques, andsilence suppression conditions, etc. Call state is maintained (stored)within the network throughout the entire pendency of the call, i.e.,between call set-up and call tear-down.

Whereas call state typically defines the nature and quality of a call,call control defines “actions” associated with the call. Call control isoften software code communicated in relation to a call. Execution of thecode accomplishes the desired action. Actions include, as examples,commands to activate a ringer, send a dial tone, detect dialed digits,and send signaling data.

Like the PSTN, components forming a VoIP network must perform four basicfunctions: (1) signaling; (2) database services; (3) call connection anddisconnection (bearer control); and (4) CODEC operations. FIG. 2conceptually illustrates the VoIP network.

For purposes of this description, the IP Backbone can be viewed as onelogical switch. The logical switch is, however, a distributed system.The IP backbone provides connectivity amongst a great number ofdistributed elements. Depending on the VoIP protocols used, this systemas a whole is sometimes referred to as a softswitch architecture.

Signaling in a VoIP network is just as critical as it is in the legacyphone system. The signaling in a VoIP network activates and coordinatesthe various components to complete a call. Although the underlyingnature of the signaling is the same, there are some technical andarchitectural differences.

Signaling in a VoIP network is accomplished by the exchange of IPdatagram messages between network components. The format for thesemessages is covered by a number of standards, or protocols. Regardlessof the protocol or hardware components used, the messages are criticalto the function of a voice-enabled IP network and need special treatmentto guarantee their delivery across the IP backbone.

Among other functions, database services are a way of locating anendpoint and translating an address communicated between two (usuallyheterogenous) networks linked by the VoIP backbone. For example, thePSTN uses phone numbers to identify endpoints, while a VoIP networkmight use a Universal Resource Locator (URL) or an IP address with portnumbers to identify an endpoint. A call control database contains thenecessary mappings and translations to identify endpoints. Functionallysimilar to call state and call control in the PSTN, “session state” in aVoIP network defines the nature of the call and controls the activitiesof components in the network.

The connection of a call in a VoIP network is made by two endpointsopening a communication session between each other. In the PSTN, thepublic (or private) switch connects logical DS-0 channels through thenetwork to complete a call. In a VoIP implementation, this connection isa multimedia stream (audio, video, or both) transported in real timebetween endpoints. This connection is referred to as the bearer (orpayload) channel and represents the voice or video content beingdelivered. When communication is complete, the IP session is ended andnetwork resources are released.

As already noted, voice communication is analog, while data networkingrequires digital data. The process of converting analog voice intodigital data is done by a coder-decoder (CODEC). There are many wellknown ways to convert analog voice into digital data. The processes usedto convert, compress, and otherwise manipulate voice data in digitalform are numerous and complex. Most are governed by publicly availablestandards.

The major components of a VoIP network are similar in functionality tothe components forming a circuit-switched network. VoIP networks mustperform all of the same tasks performed by the PSTN, in addition toproviding a gateway between the VoIP network and the existing PSTN.Although using different technologies and approaches, some of the samecomponent concepts that make up the PSTN are found in VoIP networks. Asillustrated in FIG. 2, there are three general components forming a VoIPnetwork: (1) media gateways 8; (2) media gateway controllers 9; and (3)the IP backbone 7.

Media gateways are responsible for call origination, call detection, andCODEC functions, including at least analog-to-digital conversion andvoice packet creation. In addition, media gateways have optionalfeatures, such as data compression, echo cancellation, silencesuppression, and statistics gathering. Media gateways form the interfaceallowing voice data to be transported across the IP network. In otherwords, media gateways are the source of bearer traffic. Typically, eachcall is a single IP session transported by a Real Time TransportProtocol (RTP) that runs over a User Datagram Protocol (UDP). Mediagateways exist in several forms, ranging from a dedicatedtelecommunication equipment chassis to a generic PC running VoIPsoftware.

Media gateway controllers house the signaling and control services thatcoordinate the media gateway functions. Media gateway controllers areresponsible for call signaling coordination, phone number translations,host lookup, resources management, and signaling gateway services to thePSTN.

The IP infrastructure must ensure smooth delivery of the voice andsignaling packets to the VoIP components. Due to their dissimilarities,the IP network must treat voice and data traffic differently. That is,voice and data traffic require different transport handlingconsideration and prioritization.

While there is correlation between VoIP and circuit-switchingcomponents, there are also many significant differences. One differenceis found in the transport of voice traffic. Circuit-switchingtelecommunications can best be described as a Time-Division-Multiplexing(TDM) network that dedicates channels or reserves bandwidth as it isneeded out of the trunk lines interconnecting an array of switches. Forexample, each phone call reserves a single DS-0 channel, and anassociated end-to-end connection is formed to enable the call.

In contrast to the circuit-switching network, IP networks arepacket-based networks based on the idea of statistical availability.Individual packets may be transported via a multiplicity of differentpaths through the IP network. This phenomenon often requires datapackets to stop and be stored during their journey across the VoIPnetwork. When this happens some form of queue management is requiredassess and handle the packets according to their priority indication.Accordingly, a packet prioritization schedule or Class of Service (CoS)definition is used to direct packets from specific VoIP applications inan appropriate manner. Such prioritization of IP network resourcesallows voice applications to coherently function across the IP networkwithout being adversely affected by other data traffic on the IPbackbone.

As noted above, the VoIP environment is one defined by standards. Twomajor standards bodies govern multimedia delivery over packet-basednetworks: the International Telecommunications Union (ITU) and theInternet Engineering Task Force (IETF). These two organizations areresponsible for multiple protocols and equipment standards commonly usedby participants in the VoIP market. A basic understanding of existingprotocols is necessary to fully appreciate the present invention. Thesestandards will be briefly described below, but a compete description maybe readily obtained from numerous public sources including theorganizations noted above.

Signaling System Seven (SS7)

SS7 has been previously referenced in the discussion of the PSTN. FIG. 3conceptually illustrates SS7 which is a widely used suite of telephonyprotocols expressly designed to establish and terminate phone calls. TheSS7 signaling protocol is implemented as a packet-switched network. SS7is both a protocol and a network designed to signal voice services. SS7is a unified interface for the establishment of circuit-switching,translation, and billing services.

SS7 is not built on top of other protocols. Rather, as shown in FIG. 3,it is completely it own protocol suite from physical layer toapplication layer. For networks transporting SS7, it is important thatthese services be either translated or tunneled through the IP networkreliably. Given the importance of SS7 signaling, it is necessary toensure that these messages are given priority in the network. VoIPnetworks require access to the SS7 facilities in order to bridge callsonto the PSTN.

H.323

ITU recommendation H.323 specifies a packet-based multimediacommunication system. The specification defines various signalingfunctions, as well as media formats related to packetized audio andvideo services.

H.323 standards were generally the first to classify and solvemultimedia delivery issues over Local Area Networks (LAN) technologies.However, as IP networking and the Internet became prevalent, manyInternet RFC standard protocols and technologies were developed andsometimes based on H.323 ideas. H.323 networks consist of media gatewaysand gatekeepers. Gateways serve as both H.323 termination endpoints andinterfaces with non-H.323 networks, such as the PSTN. Gatekeepersfunction as a central unit for call admission control, bandwidthmanagement, and call signaling. A gatekeeper and all its managedgateways form a H.323 “zone.” Although the gatekeeper is not a requiredelement in H.323, it can help H.323 networks to scale to a larger sizeby separating call control and management functions from the gateways.

H.323 specifications tend to be heavy handed and initially focused onLAN networking. Not surprisingly, the standard has some scalabilityshortcomings. One H.323 scalability issue is its dependency on TCP-based(connection-oriented) signaling. It is difficult to maintain largenumbers of TCP sessions because of the greater overhead involved.

With each call initiated, a first TCP session is created using a firstprotocol defining a set of messages. A TCP connection is maintainedthroughout the duration of the call. A second session is establishedusing another protocol. This TCP-based process allows an exchange ofcapabilities, master-slave determination, and the establishment andrelease of media streams.

The H.323 quality of service (QoS) delivery mechanism of choice is theResources Reservation Protocol (RSVP). This protocol does not have goodscaling properties due to its focus and management of individualapplication traffic flows. For these and other reasons, H.323 hashistorically been deemed ill-suited for service provider applications,and has been relegated to enterprise VoIP applications.

Real-time Transport Protocol (RTP)

The RTP protocol provides end-to-end delivery services for data withreal-time characteristics, such as interactive audio and video. Servicesinclude payload type identification, sequence numbering, time stamping,and delivery monitoring. Further, the RTP protocol provides features forreal-time applications, including timing reconstruction, loss detection,content delivery, and identification of encoding schemes. Many mediagateways that digitize voice applications use RTP to deliver the voicetraffic. For each session participant, a particular pair of destinationIP addresses define the session in order to facilitate a single RTPsession for each call.

RTP is an application service built on UDP, so it is connectionless withbest-effort delivery. Although RTP is connectionless, it does have asequencing system that allows for the detection of missing packets. Aspart of its specification, the RTP identifies the encoding scheme usedby the media gateway to digitize voice content. With different types ofencoding schemes and packet creation rates, RTP packets can vary in sizeand interval. The combined parameters of a RTP session dictate how muchbandwidth is consumed by the voice bearer traffic. RTP transportingvoice traffic is the single biggest data contributor to conventionalVoIP traffic.

Media Gateway Control Protocol (MGCP)

The MGCP (like its predecessor SGCP) largely defines the contemporarysoftswitch architecture. It is a master-slave control protocol thecoordinates the action of media gateways. In effect, the MGCP dividesthe functional role of traditional voice switches between the mediagateway and the media gateway controller.

In MGCP nomenclature, the media gateway controller is often referred toas a “call agent.” The call agent manages the call-related signalingcontrol intelligence, while the media gateway informs the call agent ofservice events. The call agent instructs the media gateway to setup andtear-down connections when calls are generated. In most cases, the callagent informs the media gateway to start an RTP session between twoendpoints.

As one might imagine, the control and response sequence orchestrated bythe MGCP between the media gateway and the call agent requires asubstantial quantity of call state data to be stored at both the mediagateway and call agent. The signaling exchanges between call agent andmedia gateway are accomplished by structured messages inside UDPpackets. The call agent and media gateways typically have retransmissionfacilities for these messages, but the MGCP itself is stateless. Hence,a lost message is timed out by the VoIP components, as compared with aTCP delivery mechanism where the protocol attempts to retransmit a lostpacket. Accordingly, MGCP messages are usually given very high priorityin the VoIP network over other non-real-time data packets.

A number of routine MGCP functions executed during a call will bedescribed in relation to FIG. 4. A call begins when a user picks up theorigination phone 10 and dials a destination number. The media gateway11 then notifies the media gateway controller (call agent) 12 that acall is incoming. The media gateway controller 12 looks up the dialedphone number and directs the media gateways to create a RTP connection(i.e., it identifies an IP address and port number). Media gatewaycontroller 12 also informs the destination media gateway 13 of theincoming call, and the destination phone 14 rings. Thereafter, mediagateways 11 and 13 open an RTP session across the IP network 15 whendestination phone 14 is answered.

Session Initiation Protocol (SIP)

SIP is part of the IETF's multimedia data and control protocolframework. SIP is a powerful client-server signaling protocol used inVoIP networks. SIP handles the setup and tear down of multimediasessions. Such sessions may include multimedia conferences, telephonecalls, and multimedia content deliveries.

SIP is a text-based signaling protocol transported using either TCP(transmission control protocol) or UDP (user datagram protocol). SIP isrelatively simple, efficient, and extendable, owing much of its designphilosophy and architecture to the Hypertext Transfer Protocol (HTTP)and Simple Mail Transfer Protocol (SMTP). Thus, SIP uses invitations tocreate Session Description Protocol (SDP) messages to carry outcapability exchange and setup call control channel use. Such invitationsallow “participants” to agree on a set of compatible media types.

SIP supports user mobility by proxying and redirecting requests to theuser's current location. Users may inform the server of their currentlocation (an IP address or URL by sending a registration message to aregistrar. Such a capability is increasingly attractive given a largeand growing base of mobile users. With its mobile features, SIPimplementations tend to be more discrete and SIP clients tend to belarger in number and more geographically distributed. Hence, one of thegreat challenges to implementing SIP services is mapping CoS deliveryfor the signaling and bearer traffic.

BRIEF SUMMARY OF THE INVENTION

The PSTN was designed decades ago. Its infrastructure costs too much andis not capable of delivering the telecommunication services demanded bytoday's customers. That is, Digital Loop Carriers (DLCs) and CLASS 5Switches were designed to deliver narrow-band voice using TDMtechnology. Today's customers expect data, streaming audio and video,and sophisticated features in addition to basic voice service. TDMequipment simply cannot support the modern packet-based transmissiontechnologies required to provide these services.

Unfortunately, the capital investment made by service providers in thePSTN is immense. Thus, it is reasonable to expect that large portions ofthe PSTN will remain in use, at least in some form, for many years tocome. VoIP networks must therefore continue to address the issue ofintegration with the PSTN. Stated another way, successful VoIParchitectures must inter-operate with the PSTN, yet provide a clearmigration path away from conventional PSTN components to evolving VoIPcomponents.

The present invention provides just such an architecture. Within thedictates of the present invention, distributed packet technology enablesthe deployment of new, high-margin services to customers. Yet, it alsoallows VoIP components to be added to existing networks incrementallyand transparently. The present invention provides a completelytransparent bridge between a VoIP network and the PSTN. That is, a useroriginating a call on the PSTN will not perceive in any manner that thecall was actually carried by a VoIP network. Similarly, a useroriginating a call through a media gateway will not perceive that thecalled party is accessed by the PSTN.

This objective is achieved by the present invention in the provision ofa signaling gateway that functions on a transaction-by-transactionbasis, and in the provision of a competent access switch more rationallydefined in functional relation to the signaling gateway. In contrast,conventional VoIP components implement signaling on a call-by-callbasis. Because of the call-by-call signaling approach taken and with thehistoric functional boundaries established between conventional VoIPcomponents, session data is inefficiently stored and utilized.

Several technologies have recently converged to enable the VoIParchitecture provided by the present invention. First, the ability totranslate a variety of subscriber interfaces, including analog voice,into packet streams using Digital Signal Processing (DSP) has matured tothe point where DSP hardware and software are available as reasonablypriced commodities. Second, support for Quality of Service (QoS) in thetransport of delay sensitive traffic, like voice and video, is availablein many contemporary network components. Third, a new signalingprotocol, SIP, has been developed which addresses the needs of the PSTNand efficiently facilitates the implementation of new services enabledby new, high-bandwidth transport networks.

Utilizing the capabilities of these enabling technologies, the presentinvention addresses the concept of media translation, generalizes it ina novel fashion, and solves the lingering problem of session statemanagement that has to date precluded the cost-effective scaling up ofVoIP (softswitch-based) architectures.

The novel media translation concepts underlying aspects of the presentinvention result in a new partitioning of session state between VoIPcomponents. In brief and expressed in terms of conventional components,the present invention “pushes” session state from a signaling gateway(historically a media gateway controller) into an access switch(historically a media gateway). The quantity and character of thesession state being pushed from media gateway controller to mediagateway will vary by equipment vendor.

In part, it is the transfer of session state from signaling gatewaycomponent to access switch component that allows the functional statemachine of signaling gateways included within VoIP networks designedaccording to the present invention to operate on atransaction-by-transaction basis, rather than a call-by-call basis. A“transaction” is broadly defined as including at least call setup, calltear-down, and all feature invocations. This approach to session statestorage and management within a single access switch component alsoallows VoIP networks designed according to the present invention to bereadily scaleable and beneficially susceptible to the introduction ofnew features.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a conceptual illustration of major components forming thepublic switched telephone network (PSTN);

FIG. 2 is conceptual illustration of major components forming aconventional Voice over Internet Protocol (VOIP) network;

FIG. 3 illustrates the elements of the Signaling System Seven;

FIG. 4 illustrates functional relationships between a conventional mediagateway and media gateway controller;

FIG. 5 illustrates the role of a conventional media gateway and mediagateway controller in bridging an IP network and the PSTN;

FIG. 6 is a conceptual illustration of one embodiment of a VoIP networkaccording to the present invention;

FIG. 7 is block diagram illustrating the access switch shown in FIG. 6is some additional detail;

FIG. 8 conceptually illustrates the ability of a VoIP networked formedin accordance with the present invention to allow for incrementalmigration of existing networks;

FIG. 9 further illustrates one embodiment of the signaling gatewaycomponent shown in FIG. 8;

FIG. 10 further illustrates another embodiment of the signaling gatewaycomponent shown in FIG. 8;

FIGS. 11 and 12 illustrate the flow of signaling messages in anexemplary scenario;

FIGS. 13 and 14 illustrate the flow of signaling messages in anotherexemplary scenario; and,

FIGS. 15 and 16 illustrate the flow of signaling messages in yet anotherexemplary scenario.

DETAILED DESCRIPTION OF THE PRESENTLY PREFERRED EMBODIMENTS OF THEINVENTION

Conventional VoIP architectures are largely based on two elements, theMedia Gateway Controller (MGC), sometimes referred to as a softswitch,and the Media Gateway (MG), also referred to as a trunking gateway. FIG.5 illustrates the relationship of Media Gateway 20 and the Media GatewayController 21 as components bridging a VoIP network 22 and the PSTN 23.The functional roles of these two components have previously beendescribed. In short, the combination of a MG and a MGC can be used tomove packetized payload and signaling data between VoIP network 22 andPSTN 23. Media Gateway 20 can be used to convert packetized voice datainto TDM voice for transmission over the PSTN. Media Gateway Controller21 can be used both to control operation of Media Gateway 20 andinter-operate with the signaling components of the PSTN.

Unfortunately, the conventional combination of a MG and a MGC suffersfrom a number of difficulties, including poor (or weak) signalingprotocols, and scaling limitations associated with the basicarchitecture and functional partitioning between the elements. Most MGCsare designed to interface with their MGs using hierarchal protocols likeMGCP and SGCP. These “hierarchical” protocols have been brieflydescribed above and are quite specialized in their application. Theycompare very unfavorably with “peer-to-peer” protocols like SIP in thatthey are difficult to generalize and not readily adaptable to thecontrol of components other than conventional MGs. Further, the legacyequipment-driven architecture that defines the MGC/MG functionalboundary is inherently wasteful. Since a substantial duplication ofsession state is found in both the MGC and MG, hardware resources areinefficiently utilized. These two factors greatly limit the usefulnessof conventional VoIP components in the deployment of new voice servicesand features.

In one aspect, the present invention overcomes these limiting factors byproviding a readily scaleable architecture, along with enablingcomponents and methodologies, that are easily adapted to theimplementation of new voice features and services. FIG. 6 conceptuallyillustrates one embodiment of this new VoIP architecture. Theintegration of this VoIP network with existing networks, like the PSTN,will be described later.

As illustrated in FIG. 6, the present invention proposes a VoIParchitecture comprising access devices 30, access switches 31,application servers 32, and at least one network manager 33 connected tothe IP backbone 22.

Access devices can be any device capable of receiving and/ortransmitting communications. They can be as simple as a traditionaltelephone connected via a twisted line pair, or as complex as a personalcomputer with a broadband connection or high-resolution television.Additional examples of access devices 30 include cell phones, Internetphones, computers and workstations, Personal Digital Assistants (PDAs),and portable audio playback devices such as MP3 players.Architecturally, access devices may be generally viewed as devicesfocused on one or more specific user application(s), e.g., voice, data,or video. Interfacing is the primary communications function associatedwith these access devices. Integrated Access Devices (IADs) are alsocontemplated in the term “access device(s).” IADs collect severalindividual subscriber interfaces for transmission via an aggregate trunkinterface. For purposes of the present invention, the IAD trunk uplinkis treated similarly to any other individual access interface.Similarly, the proposed access switch is capable of connecting manytypes of trunking interfaces as if they were an access device. Suchtrunking interfaces include Cable Television (CATV) systems, andwireless telephone and data interfaces, including the newer 2.5 G and 3G elements.

As presently preferred, the IP Backbone 22 shown in FIG. 6 is an IP/MPLS(Multi-Protocol Label Switching) core. IP/MPLS routers provide theimplementation of core network data transport. These conventionalelements represent a mature packet transport infrastructure. IP/MPLSrouters are assumed to provide high-speed transport only. They lack anynotion of session state. A primary requirement for the IP/MPLS routersis the support of QoS. When IP is used without MPLS, the IPType-of-Service (TOS) or Differentiated Services priority field may beused to provide multiple Classes of Service (CoS). Service LevelAgreement (SLA) QoS becomes easier when MPLS and Virtual Private Network(VPN) technologies are added. Obviously, there are numerous optionsavailable to those of ordinary skill in the art selecting a particularIP backbone, but the use of IP/MPLS is presently preferred sincecarriers are investing immense resources to further build out IP/MPLSbased data transport networks.

The component shown in FIG. 6 as Application Server 32 represents abroad class of conventional devices and systems that provide content.“Content” can be any type of electronic media, including but not limitedto audio, video, email and voice mail. The present invention allows forall types of content and application servers.

The Network Manager component 33 shown in FIG. 6 conceptually representsone or more mechanisms or systems by which all other elements in theVoIP network are controlled. In effect, network management component 33provides a window into the operation of the overall network.

More specifically, the network management component preferred by thepresent invention provides a variety of services includingconfiguration, provisioning, traffic engineering, and billing support.It must inter-operate with a carrier's existing Operations SupportSystem (OSS). The present invention contemplates the use of one or moreopen standards such as SNMP, COBRA, and TL/1 in cooperation with accessswitch 31 and a signaling gateway (described below). The networkmanagement component preferably provides an intuitive, graphicalinterface allowing quick appraisal and manipulation of the networkelements. Those of ordinary skill in the art are well acquainted with abroad class of conventional technologies and vendors providing networkmanagement tools and systems. Since the architecture proposed by thepresent invention consists of many small switching elements anassociated network management component should be easily scaleable.

The design and functional operation of access switch 31 are importantaspects within the architecture proposed by the present invention.Access switch 31 provides at least access device termination, sessionstate maintenance, and packet switching capabilities to the VoIPnetwork. Access switch 31 is expected to implement an interface for eachconnected access device 30. Given the number, breadth, and ever growingnature of potential access devices this requirement may appear dauntingat first glance. However, the ongoing development of standard interfacesand the modular nature of the access switch, described hereafter,greatly simplify this effort.

Access switch 31 maintains session state associated with terminatingsessions being carried over subscriber interfaces. The term “subscriberinterface” means any reasonable connection between an access device,integrated access device, or trunking interface and the access switch.In the context of a voice call, the term “session state” includes atleast call state and call control information. Unlike conventional VoIPelements, the access switch according to the present invention isadapted to maintain all session state. The combination of access devicetermination and session state maintenance within a single networkcomponent provides a natural environment for feature and serviceimplementation. Feature implementation may be readily performed in itsentirety at the access switch, since a complete record of session stateis available. Integration of packet switching and QoS support allowsdeterministic and flexible content support.

The access switch described by the present invention addresses most ofthe complexity associated with handling communications in the proposedVoIP architecture. By pushing complexity into the access switch, acomponent operating at the very edge of the VoIP network, the presentinvention enhances overall network reliability by reducing the densityof each individual component. Scalability is further enhanced since theamount of work required of the network core is reduced. Additionally,feature and service creation can be readily monitored and maintained bythe carrier. Thus, the location of system complexity and featuredefinition capabilities in peripherally located access switches providesgreater system security and frees system subscribers from the need tohandle of day-to-day configuration details.

FIG. 7 is a block diagram showing one presently preferred embodiment ofaccess switch 31 is some additional detail. In this embodiment, accessswitch 31 comprises one or more line interface cards 40 that “terminate”access devices and maintain session state, an IP/MPLS backplane 41allowing packet switching between the access switch elements, and packetinterface cards 42 that perform routing and forwarding between theIP/MPLS trunk interfaces. The packet interface cards may be redundantlyprovided. The development of line interface cards and definition of datapacketization in relation to specific subscriber interfaces and accessdevices is deemed an exercise in ordinary skill. The specific definitionof the packet interface cards is similarly deemed ordinary. Manyapproaches to the definition and implementation of a high bandwidthbackplane will be successful within the context of the presentinvention. Because high bandwidth IP/MPLS transport is used across itsbackplane, access switch 31 is readily able to accept future subscriberinterfaces by simply adding new line interface cards into the existingchassis.

As presently preferred, the standard, rack-mounted chassis used to houseaccess switch 31 contains no common equipment. Rather, all of thefunctions traditionally associated with common equipment are distributedonto line interface cards 40 and packet interface cards 42. Such anapproach provides lower entry cost for the access switch and allows itto be used in low density installations at cost points similar to thoseof high density installations.

While TDM and ATM are in prevalent use by conventional networkequipment, IP and MPLS based packet switching equipment will dominatethe networks of the future. The access switch must be capable ofswitching packet data onto a carrier's network regardless of the datatransmission scheme being used. One principle challenge in this contextis maintaining QoS for packets carrying data associated withtime-sensitive applications, like voice and video. The access switchaccording to the present invention distributes packet switching acrossboth line interface and packet interface cards. Data transmissionbetween the line and packet interfaces occurs at line speed. Thus, nostore-and-forward buffering is used until packets arrive at the packetinterface card. Since no congestion can occur prior up to the packetinterface card, QoS implementations can be localized. UsingDifferentiated Services (Dissserv) in conjunction with the proposedaccess switch allows a carrier to provide up to eight classes of packettransport service to and from a trunking interface.

As presently contemplated, the access switch is preferably connected toa VoIP network supporting IP/MLS bearer transport and SIP signaling. Itis fully expected that SIP will evolve over time and will almostcertainly be replaced in practice by some yet to be defined protocol.While use of SIP is presently preferred, those of ordinary skill in theart will understand that the present invention contemplates use of anyexpressive, peer-to-peer protocol over an IP network. Thus, in oneaspect, the present invention may be characterized as using a selectedpeer-to-peer protocol to facilitate transparent cross network exchangesbetween IP networks and non-IP networks, such as the PSTN or an H.323zone.

As already noted, a primary function of the proposed access switch isthe termination of access devices. Access devices are terminated via oneor more subscriber interfaces. The term “terminate” or “termination”means translating both the payload data and signaling (if any)associated with a subscriber interface into a common representation usedfor transport via a VoIP network. A primary advantage of terminating asubscriber interface at an access switch is flexibility. The accessswitch can be designed to terminate a variety of different subscriberinterface types, thereby allowing a carrier to service a variety ofcustomers using a single network component.

In the presently preferred embodiment, as illustrated in FIG. 7, accessdevice termination is accomplished by individual line interface cards40. Since line interface cards 40 are removable and replaceable, theaccess switch may be readily adapted to different subscriber interfaces.A variety of line interface cards 40 terminating local telephonecarriers have been developed including; a 32-port Asymmetric DigitalSubscriber Line (ADSL)/Plain-Old Telephone Service (POTS) line interfacecard and a 16-port T-1 line interface card. In presently preferred lineinterface cards, all of the POTS voice data is encoded using one of avariety of DSP CODECS, including G.711, G.723, G.726, G.729. Inaddition, T.38 FAX and V.92 modems can also be terminated. The ADSL lineinterface card may support both full and G.Lite implementations. The T-1interfaces can be configured as both Channelized, supporting CAS, PRI,and TR-008 signaling and Un-Channelized, supporting Frame Relay service.

In addition to connecting to the IP backbone and terminating accessdevices, the access switch of the present invention is able to maintainall session state. Conventionally, session state is maintained by the MGand MCG during the entire session. For voice calls, session state musttherefore be maintained from call setup, during call “hold time,” untiltear-down. Hold time is the period of time during which the voiceconversation is carried out.

The access switch according to the present invention maintains allsession state associated with a subscriber entirely on one or more lineinterface cards to which the subscriber interface is attached. Like theconventional combination of MG and MGC, session state is maintained onthe access switch throughout the entire call. However, unlikeconventional VoIP networks session state is maintained on a singlecomponent, not two separate components often manufactured by differentvendors. Session state may include feature and services implementation,as desired. Consider, for example, certain features generally associatedwith a CLASS 5 switch, namely a calling name lookup associated with thedisplay of caller ID information or local number portability initiatedby call manager software on a line interface card associated with thespecific subscriber interface. These features can be implemented in theaccess switch, thereby beneficially offloading the conventionalnarrow-band equipment.

The ability to maintain all session state on the access switch is adistinct difference between the VoIP architecture proposed by thepresent invention and conventional VoIP architectures. Maintaining allsession state on the access switch eliminates the need for theconventional MGC. In conventional VoIP network architectures, mostsession state is maintained at the MGC which typically resides at acentral office location. A major problem with conventional VoIParchitectures and implementations is the fact that the MGC cannotmaintain all session state in a single location, because the MGhistorically requires that some session state be stored in the MG. Thisrequirement results in a detrimental duplication of session stateinformation. While some benefit may be identified in centralizing largeportions of session state, the cost associated with this advantage arevery high due to the added burdens of redundantly storing session statedata and maintaining coherency between session state data portionsseparately stored at the MG and MGC.

In addition, duplication of session state makes feature implementationmore difficult. In most cases, conventional feature implementationrequires some interaction between MG and MGC. In practice, the MG andMGC are often manufactured by different companies. This unfortunatereality requires a would-be feature designer to deal with multipleequipment vendors to order to define or redefine a feature. However,when all session state is stored in a single network component, asprovided by the architecture proposed in the present invention, featureimplementation is greatly simplified, both conceptually and in practice.

Session state maintenance is one principle driver for the VoIParchitecture proposed by the present invention. The economic realitiesof equipment migration is another principle driver. As noted, anysuccessful VoIP architecture must provide an efficient path forequipment migration while also providing effective integration withexisting networks like the PSTN. Migration must be both incremental andtransparent. The proposed architecture, along with its enabling systemcomponents and methodologies, provides these two capabilities.

The concept of transparent interoperability that underlies successfulequipment migration is illustrated in the context of the presentinvention by the system schematic shown in FIG. 8. (Compare FIGS. 5 and6, in which like elements are similarly numbered). Here, the combinationof a signaling gateway 34 (described below) and access switch 31 is ableto replace the conventional combination of media gateway 20 and mediagateway controller 21. The combination of signaling gateway 34 andaccess switch 31 is not just a physical, one-for-one replacement for theconventional combination, but a combination allowing enhancedscalability of the proposed VoIP network at reduced cost.

The concepts underlying the design of signaling gateway 34 weredeveloped in relation to the proposed access switch 31. For example,because access switch 31 maintains all session state, signaling gateway34 need not maintain any session state. At most, signaling gateway 34must maintain “transaction state” information during the very briefperiods of time required to execute an associated transaction. Whiletransaction state is sometimes considered a subset of session state, thetwo types of state data are considered distinct for purposes of thepresent description. Session state is all state typically persistingbeyond the transaction periods. In contrast, transaction state is stateexisting in a signaling gateway only during the pendency of thetransaction.

Signaling gateway 34 provides transparent translation between twosignaling networks. In the context of the working example used thus farto illustrate the present invention, “transparency” allows a signalinggateway which provides the illusion that the SIP signaling network is anextension of the SS7 network, and vice versa. Thus, where access switch31 is used to bridge two networks having different signalingenvironments, signaling gateway 34 becomes essential to transparency. Onthe other hand, it is the ability of access switch 31 to maintain allsession state that allows signaling gateway 34 to operate in a statelessmanner.

FIG. 9 further illustrates one presently preferred embodiment ofsignaling gateway 34 in the context of bridging an IP network 50 usingSIP with a non-IP network like the PSTN 23 which relies on the SS7network for signaling. At a first port 35 (which may actually consist ofnumerous, separate data connections), SIP messages are sent and receivedfrom the IP network 50. At a second port 36 (again possibly formed bynumerous, separate data connections), SS7 signaling messages are sentand received from the PSTN 23. SIP messages are generated and parsed bya SIP parser/generator 37. A conventional SIP message parser/generators(commercially available) may be used here. SS7 signaling messages aregenerated and decoded by a SS7 protocol stack 38. Again, one of severalconventional devices may serve here. Commercially available SS7 protocolproducts maintain much of the state information used to establish thereliability and redundancy of the logical SS7 connection to the PSTN.

Functionally connecting SIP parser/generator 37 and SS7 protocol stack38 are SIP transaction handler and translator 39. These two elements areconceptually illustrated as one element since together they form the“glue” binding the SIP and SS7 networks together. The SIP transactionhandler provides termination for the SIP branch of the signaling path.The translator provides a “simple” translation between SIP messages andanalogous SS7 signaling messages.

Three basic principles motivate the design and implementation of thesignaling gateway and the translator central to its purpose. First,translation should occur by direct mapping of message types from onesignaling network into messages types from the other signaling network,wherever possible. This so-called “direct translation” is desirablebecause it takes advantage of all natural mappings between the semanticsof the two signaling networks. It also makes the translator as simple aspossible. For example, there are numerous analogous signalingtransactions in SIP and SS7, most specifically in the area of call setupand tear-down.

Second, the signaling gateway should be, functionally speaking, only asimple translator. In other words, if an exception (error) conditionoccurs, the signaling gateway does nothing to correct the condition. Anindication of an exception is merely translated and passed through inexactly the same way as any normal signaling message. It is theresponsibility of the signaling endpoints to handle exceptions. Morerobust system implementations are possible by allowing the signalinggateway to pass exceptions along without attempting to handle them. Thesignaling gateway thus avoids complexity and the need to observestate-based functionality requirements.

Third, in those situations where one signaling network is more“expressive” (i.e., has more functions) than the other signalingnetwork, mechanisms (hardware and/or software) must be added to the lessexpressive signaling network to order to allow as much direct mapping aspossible. Staying with the working example, some existing SS7 signalingmessages do not directly map into SIP messages.

Adding mechanisms to one set of signaling representations in order toallow the mapping of a feature already present in one network to appearin another network can be difficult. Yet there are several approachesavailable to address this problem. One approach develops permanentadditions to the set of signaling representations lacking a particularfeature. In the presently preferred embodiment, new SIP messages can beadded to the existing protocol. Alternatively, a generic transportmessage could be used with the protocol content contained in the SIPmessage body. This later approach is presently preferred. While theformer approach has the advantages of forming high quality constructs inSIP to augment the existing message set, it also requires that all suchmessages be provided in every SIP implementations, which is clearly notnecessary. The later approach has the advantage of decoupling PSTNspecific functions from the generic SIP protocol, which has manyadditional uses beyond PSTN interoperability.

The translator resident in the signaling gateway performs messageconversion on a transaction by transaction basis. As already noted,transactions include call setup, call tear-down, and feature invocation.The translator element maintains a transaction control block (i.e.,transaction state) for each transaction from only the time thetransaction begins until the transaction is completed.

Maintaining minimal transaction state in the signaling gateway over thevery brief period of time required to execute the transaction, ratherthan maintaining session state during an entire call is important fortwo reasons. First, transactions are always shorter than a call, oftenvastly shorter. A typical transaction may last for a few hundredmilliseconds. In contrast, a call may last hours. By briefly maintainingonly transaction state, the signaling gateway makes much more efficientuse of available resources. Second, error handling for signalingprotocols tend to be based on errors occurring during a transaction. Ifsession state is maintained between transactions, the signaling gatewaymust assume some responsibility for error handling outside of protocoldefinitions. This greatly complicates the design of the signalinggateway and its operation with the access switch. Historically, thiscomplex interaction between the signaling element (an MGC) and theswitching element (a MG) necessitated the development and use of acompletely separate, master-slave protocol (MGCP). By only maintainingtransaction state during transaction periods, the signaling gatewayaccording to the present invention need not handle errors between callsetup and tear-down. In the present invention, this responsibility ispassed in its entirety to the access switch which, maintaining allsession state and being located at the periphery of the network, is muchbetter suited to resolve such issues.

In addition to providing a signaling gateway capable of transparentlytranslating messages between networks using different signaling schemes,signaling gateway 34 of FIG. 9 may be combined with a competent server.In the preferred embodiment, a full SIP server is contemplated. The SIPserver may be a conventional, off-the-shelf device, but should includesupport for all, current, standard features including at least proxying,redirection, routing, and forking. The SIP servers (or SIPfunctionality) incorporated into the present invention is expected toconform with SIP standards, such as those set forth in Handley et al.,RFC 2543, Session Initiation Protocol available through the IETF.

Conventional SIP servers include a parser receiving data from a bufferednetwork connection. The parser assembles received data bits intocomplete SIP messages. Once assembled by the parser, complete messagesare forwarded to a routing engine that routs the incoming messagesaccording to several well understood criteria. Conventional SIP serversalso include a generator receiving SIP messages in their internalrepresentation, converting the internal representations into SIP messagetext, and transmitting the message text as a stream of data via anetwork connection.

A second embodiment of a signaling gateway 60 is illustrated in FIG. 10.Here, one or more signaling gateways 34, like the one described inrelation to FIG. 9, are combined with one or more SIP servers 59. Eachsignaling gateway 34 communicates with the SS7 signaling network. EachSIP server communicates with one or more external devices, typically anaccess switch 31, but possibly other devices such as the conventionalMGCs 21. The SIP servers 59 and signaling gateways 34 are preferablyindependent. This means each can execute on the same hardware platformor on separate hardware platforms. Multiple SIP servers 59 can interactwith multiple signaling gateways 31. The use of SIP messages ispresently specified for communication between the SIP server andsignaling gateway elements. The use of a standard representation forthese communications is desirable, but a more efficient representationmay be used if performance dictates.

Continuing with the working example of bridging a SIP network and a SS7network, a variety of signaling must pass between these two networks inorder to connect a call. However, all of this signaling may be logicallyviewed as being in one of two classes: 1) call setup and tear-down, or2) feature invocation. The parameters of the first class are wellunderstood and fairly limited. The second class is more elastic andbound to change over time and in relation to the access devices beingconnected to associated access switches or MGCs. Not all featureinvocations require signaling interactions across network boundaries.However, some features do, including as examples, local numberportability, calling number delivery, calling name delivery, visualmessage waiting indication, stutter dial tone message waitingindication, abbreviated ring message waiting indication, callingidentity delivery on call waiting, calling identity delivery blocking,call forwarding, and calling number delivery blocking. These exemplaryfeatures all require the use of TCAP to query SS7 SCPs.

Four possible signaling scenarios will be examined. The first is onewhere the calling subscriber and the called subscriber are both directlyconnected to respective access switches in the VoIP network. In thisscenario, no SS7 signaling is required. Thus, no message translation isrequired. The connecting access switches function as if they werecomponents in a conventional VoIP network. A signaling gateway of thesort shown in FIG. 10 might become involved in the call if an associatedSIP server is used to proxy or redirect the call. In any event, thisfirst scenario does not implicate the more novel aspects of the presentinvention.

A second calling scenario is illustrated in FIG. 11. In this scenario,the calling subscriber is connected to an access switch 31 and thecalled subscriber is connected via a conventional media gateway 21. (Inthis example and several to follow the access switch 21 and signalinggateway 34/60 are shown as being physically separate. Clearly these twoelements may, but need not be commonly located. The relationship ofthese components to the IP network and other connected networks is morerelevant than physical proximity).

FIG. 12 illustrates the flow of signaling between the network componentsshown in FIG. 11 using a call setup and tear-down example. Of note,access switch 31 and signaling gateway 34 or 60, (as shown in FIGS. 9and 10 respectively), are able to interact with the conventionalcombination of a MGC 20 and MG 21 across the IP network to effecttransport of payload and signaling data. Upon call inception accessswitch 31 issues an INVITE signal to MGC 20. MGC 20 sets up the call onthe MG using a protocol like MGCP. The MGC is also responsible forsignaling the SS7 network. It issues an IAM and awaits an ACM and asubsequent ANM that indicate the PSTN leg of the call has beenestablished. A 200 OK result is then returned to access switch 31 thatthen sends an ACK to the MGC 20. When the call is complete, accessswitch 31 issues a BYE to MGC 20, and MGC 20 tears down the call on MG21 using MGCP. It also issues a REL signal to the SS7 network and awaitsan RLC. When the RLC is returned, MGC 20 responds to access switch 31with an ACK indicating the call has been torn down.

FIG. 13 illustrates a third scenario in which both subscribers areconnected to the IP network via access switches. However, in thisscenario one subscriber is directly connected via a first access switch31 a and the other subscriber is connected via the PSTN 23 to a secondaccess switch 31 b, perhaps via a T-1 interface. Message translationbecome necessary.

FIG. 14 illustrates the flow of signaling between the network componentsshown in FIG. 13 using a call setup and tear-down example. First accessswitch 31 a issues an INVITE to second access switch 31 b. Second accessswitch 31 b allocates a T-1 channel and issues an INVITE to signalinggateway 34/60. Signaling gateway 34/60 translates the INVITE to an IAMand forwards it to the SS7 network. The SS7 network indicates theestablishment of the PSTN leg of the call by returning an ACM and an ANMmessage to signaling gateway 34/60. (Some additional intervening messagesequences that occur here on the SS7 side are omitted for the sake ofclarity). When the PSTN leg of the call is completed, signaling gateway34/60 issues a 200 OK to second access switch 31 b. Then second accessswitch 31 b sends a 200 OK to first access switch 31 a. Call setup iscomplete when first access switch 31 a sends an ACK to second accessswitch 31 b. Call tear-down proceeds similarly as shown in FIG. 14.

Taken together FIGS. 15 and 16 illustrate a fourth scenario in which aSIP component must retrieve data from a database (SCP) located withinthe SS7 network. This scenario might occur where a particular feature orTCAP service is being invoked from a SIP component. Again, messagetranslation become necessary.

The scenario begins when access switch 31 formats a text representationof a TCAP service call in a SIP INFO message and sends it to signalinggateway 34/60. Signaling gateway 34/60 translates the TCAPrepresentation in the INFO message into an SS7 TCAP invoke message andsend it to an SCP 70. SCP 70 services the request and returns a Responsemessage to signaling gateway 34/60. Signaling gateway 34/60 convertstranslates the TCAP Response into a text representation and encapsulatesit into another SIP INFO message. The feature invocation is completedwhen the encapsulated TCAP Response is returned to access switch 31.

As illustrated by the exemplary scenarios, signaling message translationmay or may not be necessary. The access switch according to the presentinvention may be required to interact with other, fully compatibleaccess switches, with conventional VoIP components, or directly with thePSTN. These scenarios are only examples. One of ordinary skill in theart may well extrapolate these teaching to connections between IP basednetworks and other non-IP based networks, such as H.323 zones.

The architecture and the enabling components and methodologies proposedby the present invention greatly enhance reliability in VoIP networks.Reliability can be viewed from two angles. The first is overall networkreliability. Carriers want their subscribers to be able to establishsessions at any time from any point within the network. For this tohappen, redundant communication paths are established, whereby thefailure of any one path does not destroy network connectivity. IP/MPLSnetwork transport is well positioned to provide carriers withunprecedented network reliability at extremely reasonable costs.

Second, reliability is viewed by individual subscribers. Over and abovethe inherent reliability associated with his access device, a subscriberperceives network reliability in terms of the frequency with which thenetwork is unavailable to his access device. In the proposedarchitecture, services are delivered by a number of network componentsworking in combination. However, reliability for a given subscriber islargely defined by the reliability of the access switch connecting hisaccess device. The proposed access switch is designed with highreliability in mind. Session state and subscriber information may bedistributed onto multiple line interface cards using well knowntechniques to minimize the overall impact of a single line interfacecard failure.

The present invention also provides enhanced scalability. Scalability isa concept applicable and important to carriers. Simply put, ascalability metric specifies how many individual users can make use ofthe carrier's network simultaneously. The two most scaleable networkarchitectures in existence are the PSTN and the Internet. It is nocoincidence that both of these networks implement a majority of their“intelligence,” i.e., feature definition and service creation, at theirrespective edges.

For example, the CLASS 5 switch, a narrow-band device increasinglyill-suited to contemporary services, is the element in the PSTN to whichthe great majority of subscribers lines are connected. In many ways theCLASS 5 switch can be thought of as being on the edge of the PSTN. Whilethe Advanced Intelligent Network (AIN) includes SCPs that maintaincentralized databases, the majority of state operations are stillperformed at the CLASS 5 switch. The interior of the PSTN, made up of amyriad of tandem switching elements are highly specialized, performinglarge-scale transport functions. The core maintains state only on alimited basis.

By pushing the session state maintenance, as well as feature and servicecreation into the access switch in the proposed VoIP architecture, thepresent invention derives enhanced scalability. This partition ofintelligence and functionality is sharply at odds with conventional VoIPnetworks which rely on specialized command protocols between mediagateways and media gateway controllers.

The architecture proposed by the present invention provides carrierswith additional control over the creation of services and addedsecurity. In order to create new or expanded services, carriers mustmaintain access and control over session state information. By movingall session state to a component at the periphery of the network andunder the control of the carrier, the present invention allows greateraccess to and control of session state. Physical security is alsoenhanced. While it is certainly possible to remotely attack a componentstoring session state, physical access to the component greatlyincreases the options available to a would-be hacker. By maintainingphysical control of the component storing all session state the securityissues confronting a carrier are markedly reduced.

The present invention has been explained in relation to severalpresently preferred embodiments. These embodiments are only workingexamples. Certain specific protocols, technologies, and components havebeen chosen as part of the examples. The invention, however, is notlimited to these examples. Rather, those of ordinary skill in the artwill readily perceive adaptations, modifications, and extrapolations oftheses teachings which fall within the scope of the present invention asdefined by the attached claims.

1. A system comprising: an access switch configured to operate at anedge of a Voice over Internet Protocol (VoIP) network, the access switchconfigured to perform packet switching, the access switch terminating anaccess device and connecting the access device to the VoIP network,wherein all session state associated with a call from the access deviceis maintained in the access switch from call setup until call tear-down,the session state comprising call state information and call controlinformation for the call; and a signaling gateway configured to connectthe VoIP network and a non-IP network, wherein only transaction stateassociated with the call and not the session state is maintained in thesignaling gateway; wherein transaction state for a transactionassociated with the call is maintained in the signaling gateway onlyduring pendency of the transaction, wherein the pendency of thetransaction is less than pendency of the call.
 2. The system of claim 1,wherein the non-IP network comprises a Signaling System Seven (SS7)network and the signaling gateway further comprises: a SIP parserreceiving SIP messages from the VoIP network; a SS7 protocol stackreceiving SS7 messages from the SS7 network; a translatorreceiving/transmitting SIP messages from/to the SIP parser, andreceiving/transmitting SS7 messages from/to the SS7 protocol stack,wherein the translator converts SIP messages from/to SS7 messages. 3.The system of claim 1, wherein the signaling gateway comprises atranslator configured to translate messages between the VoIP network andthe non-IP network.
 4. The system of claim 1, wherein the access switchfurther comprises: a line interface card terminating the access deviceand defining a subscriber interface for the access device; a packetinterface card connected to the VoIP network; and a backplane commonlyconnecting the line interface card, the packet interface card, and theVoIP network.
 5. The system of claim 1, wherein the non-IP networkcomprises a Signaling System Seven (SS7) network and the signalinggateway further comprises: a SIP server element connected to the VoIPnetwork; and a signaling gateway element translating SIP messagesreceived from the VoIP network into SS7 messages, and translating SS7messages received from the SS7 network into SIP messages.
 6. The systemof claim 1, wherein the non-IP network comprises a Signaling SystemSeven (SS7) network and the signaling gateway further comprises: aplurality of SIP server elements connected to the VoIP network; and aplurality of signaling gateway elements, each one of the plurality ofsignaling gateway elements being adapted to translate SIP messagesreceived from the VoIP network into SS7 messages, and to translate SS7messages received from the SS7 network into SIP messages.
 7. The systemof claim 6, wherein respective SIP server elements and signaling gatewayelements communicate one to another using SIP messages.
 8. A methodcomprising: terminating, by a first access switch operating at an edgeof a Voice over Internet Protocol (VoIP) network, a call from an accessdevice; determining, by the first access switch, a location identifiedby the call and identifying a second access switch associated with thelocation; maintaining all session state associated with the call in atleast one of the first and second access switches during the call, butmaintaining no session state in a signaling gateway used forfacilitating the call, the session state comprising call stateinformation and call control information for the call; and maintaining,by the signaling gateway, transaction state information associated witha transaction occurring during the call only for a time period forexecuting the transaction, wherein the period for executing thetransaction is less than a duration of the call.
 9. The method of claim8, wherein the transaction state is maintained in the signaling gatewayonly during pendency of the related transaction.
 10. The method of claim8 further comprising: facilitating, by the signaling gateway, aconnection between the VoIP network and a non-IP based network.
 11. Themethod of claim 8 further comprising: facilitating, by the signalinggateway, a connection between the VoIP network and a public switchedtelephone network (PSTN) or a H.323-based network.
 12. The method ofclaim 8 wherein the transaction is one of call setup, call tear-down, ora feature invocation.
 13. The system of claim 1 wherein the non-IPnetwork is a public switched telephone network (PSTN).
 14. The system ofclaim 1 wherein the non-IP network is a H.323-based network.
 15. Thesystem of claim 1 wherein the signaling gateway comprises a first portfor sending and receiving messages to and from the VOIP network and asecond port for sending and receiving messages to and from the non-IPnetwork.
 16. The system of claim 1 wherein the transaction is one ofcall setup, call tear-down, or a feature invocation.
 17. The system ofclaim 4 wherein the backplane uses Multi-Protocol Label Switching (MPLS)protocol.